The present invention relates to the field of sound processing circuits which handle low frequency signal components in multichannel audio signals.
With recent progress in audio signal compression technology and faster signal processing, recording and reproduction of multichannel audio signals, which have more channels than the conventional two channel stereo signals, are now being adopted in commercial equipment. Typical multichannel systems include the AC-3 system developed by Dolby Laboratories (hereafter referred to as the discrete digital multichannel system) and MPEG2. Optical disks on which audio signals are recorded employing the discrete digital multichannel system are already on the market. Decoders for converting signals recorded in the discrete digital multichannel format into ordinary signals are also available. Furthermore, at the end of 1996, software and hardware for digital video disks adopting the discrete digital multichannel system as one audio recording format were released.
The characteristics of these multichannel audio signal recording systems are (1) Audio signals for each channel can be recorded as completely independent audio signals without any correlation between channels and (2) Audio signals of a broad frequency band ranging from low frequency to high frequency, limited only by sampling frequency, can be recorded in each channel. For example, in the discrete digital multichannel system, there are five independent channels with frequency bands from 20 Hz to 20 kHz, and one channel exclusive to low frequencies up to 120 Hz.
The conventional processing method used in commercial equipment is to first encode the above multichannel audio signals and record them as 2-channel stereo signals. These stereo signals can be decoded during reproduction to reconstitute multichannel audio signals. The Dolby surround system adopts this method. This system is most frequently used for recording multichannel audio signals in movies.
The chief characteristic of this method is its feasibility to record and reproduce multichannel audio signals in a format completely compatible with two-channel stereo signals. Using this method, however, the signals in each discrete channel lose their independence since signals are produced for each channel by signal processing such as the addition and subtraction of the stereo signals recorded on the recording medium. This converts previously independent multichannel audio signals, before encoding, into completely different signals.
To reduce the above disadvantage, an active matrix circuit called the Dolby ProLogic circuit has been developed. This circuit secures the independence of each channel by reducing the sound level of the other channels when signal components of a certain channel are dominant in multichannel audio signals processed by the addition and subtraction of stereo signals, and reproducing the signals only in the dominant channel. This circuit is effective when only one channel is dominant, but much less efficient when all channels have about the same signal level.
New multichannel systems including the discrete digital multichannel system completely assure the independence of each channel during recording in the conventional two-channel stereo signal format. These new multichannel systems are used mainly for recording and reproducing sound in movies. Assurance of independence of each channel improves the clarity of spoken word, movement and direction of sound and spatial impression, allowing viewers an enhanced impression of live sound performance.
For reproducing these multichannel audio signals, speakers which can cover a broad range of frequency bands from low to high bands are preferably used. In the above active matrix system, for example, audio signals of four channels at the left, center, right and rear are decoded from input stereo signals. Audio signals for the rear channel have a frequency range from about 100 Hz to 7 kHz, and audio signals of other three channels at the left, center, and right have a broad frequency range from 20 Hz to 20 kHz.
Accordingly, it is preferable to employ the same type of speaker for at least three channels, i.e. at the left, center, and right, for covering the frequency range from 20 Hz to 20 kHz. In the above discrete digital multichannel system, it is preferable to employ speakers to cover the frequency range of 20 Hz to 20 kHz for all five channels, i.e., at the left, center, right, left back, and right back, because the signals for all five channels range from 20 Hz to 20 kHz.
However, if this type of reproduction system is introduced for home use, a large speaker for broad reproduction bands can be employed for the left and right speakers but it is generally difficult to use this type of speaker in the center because there is a display monitor for displaying video images. Also for back speakers, smaller speakers are often used due to limitations in installation space. These smaller speakers generally have less reproduction capability for low frequencies compared to large speakers.
When multichannel audio signals are reproduced in unmodified form in a system employing speakers with both good and poor low frequency reproducibility, the relative volumes of low and high frequencies may be unbalanced. The volume of low frequency sound may be insufficient if audio signals are concentrated in channels with poor low-band reproducibility. In particular, listeners may have a sense of incongruity when the sound moves from one side to the other.
To reduce these disadvantages, equipment exists which features an active matrix circuit which further employs a sound processing circuit for distributing low frequency signal components of the center channel to the left and right channels.
FIG. 7 shows an example of a sound processing circuit of the active matrix system. Audio signals input from two channels to an active matrix circuit 51 are decoded into signals for four channels: left (Lch), center (Cch), right (Rch), and back (Sch). A high-pass filter (HPF) 52 receives decoded signals for the center channel, allows through only high-band signals, and outputs them as signals for the center channel.
At the same time, signals for the center channel are input to a low-pass filter (LPF) 53. The cut-off frequency of the LPF 53 is set at almost equivalent to the cut-off frequency of the HPF 52, and it allows through only low frequency signals for the center channel. The output here is attenuated by about 3 dB by a coefficient multiplier 54, and then supplied to adders 55L and 55R for the left and right channels. The adder 55L adds the low frequency signal components of the center channel to the audio signals for the left channel, and the adder 55R adds the low frequency signal components of the center channel to the audio signals for the right channel. Consequently, these low frequency signal components are distributed to the left and right channels by the two adders 55L and 55R. The cut-off frequencies for the HPF 52 and LPF 53 are both set to about 100 Hz.
The above sound processing circuit enables the diversion of low frequency signals, originally destined for the center channel, to the left and right speakers and avoids insufficient low frequency signal components even when the center channel speaker has poor low frequency reproducibility. It is difficult to specify the position of sound source of frequency signal components lower than 100 Hz which are distributed to the left and right channels. This avoids a sense of incongruousness as to sound source direction even though the sound source is split between the left and right channels.
The active matrix circuit 51 suppresses the supply of audio signals to the center and right channels when the left channel receives large audio signals. On the other hand, when the center channel receives a large portion of audio signals, the active matrix circuit 51 suppresses the supply of audio signals to the right and left channels. This makes it unnecessary to set a surplus amplitude margin to avoid overflow of audio signals in the adders 55L and 55R which distribute the low frequency signal components for the center channel to the left and right channels.
Accordingly, the adder for distributing low frequency signal components for the center channel to the left and right channels may not require a surplus amplitude margin even when the circuit is configured using digital processing. This avoids the dropping of lower bits for securing sufficient amplitude margin. In other words, the sound processing circuit can be replaced with a digital circuit without degrading the sound quality.
In this example, configuration of the circuit is simple since it involves only distributing low frequency signal components for the center channel to the left and right channels, and therefore it is relatively easy to configure using an analog circuit. The technology employed in the active matrix circuit is further explained in detail in a range of documents such as JAS Journal (pp. 22-26, May 1989).
In case of the aforementioned new discrete digital multichannel recording and reproducing system which allows the recording of audio signals completely independently to multiple channels, the situation is slightly different.
First, since the signals for each channel are independent, the amplitude increases in response to the number of added signal components in the circuits downstream from the adder when low frequency signal components from a certain channel are distributed to other channel(s). Therefore, it is necessary to provide a surplus amplitude margin for these circuits. For example, if the low frequency signal components of a certain channel are distributed to another channel and low frequency signals for both channels have the maximum amplitude at the same phase and same level, a surplus amplitude margin of about 6 dB may be necessary in circuits downstream from the adder. If there is no such surplus amplitude margin, a 6 dB signal overflow is created in the circuits downstream from the adder.
Second, since all channels can cover signal components in broad frequency bands from low to high frequency bands, the number of channels to which low frequency signal components can be distributed increases. Accordingly, added amplitude may be converted into high-level signals. For example, if low frequency signal components for all six channels are added in equipment adopting the discrete digital multichannel system, added signals will have 6 times greater amplitude, at their peak, than the original signals. If the original signals in each channel are at 2 Vrms at the maximum, their added signals may reach 12 Vrms at their peak.
Third, an extremely complicated circuit for distributing low frequency signal components may be required to determine which low frequency signal components from which channel are to be distributed to which channel.
As described above, a sound processing circuit for distributing low frequency signal components for a certain channel to other channel(s) can be relatively easily configured by the use of a digital circuit, and its control may be facilitated. However, a large amplitude margin may be required for channels receiving distributed low frequency signal components. To secure such a large amplitude margin, the upper bits in digital signals are given priority in the sound processing circuit, resulting in the risk of lower bits in digital audio signals being dropped. The dropping of lower bits from digital audio signals may lead to degradation of sound quality.
If a sound processing circuit having the above function is configured with an analog circuit, it is relatively easy to secure an amplitude margin in channels receiving distributed low frequency signals. However, the circuit configuration may become complicated by the need to determine the distribution of low frequency signal components, and thus its control method may become more complicated.
Moreover, ordinary amplifiers which are connected downstream from the sound processing circuit often do not have any surplus amplitude margin. Overflow may occur in devices in downstream processes although overflow has not occurred in the sound processing circuit.
To avoid overload in downstream devices, an effective strategy is to provide a limiter to circuits through which audio signals pass. If the limiter is configured with an analog circuit, it results in the addition of another circuit, adding to overall circuit cost. The burden on the analog circuit may also increase because a large amplitude margin may be required for signals to be supplied to the limiter.
Furthermore, with recent improvements in the performance of digital processors, spare processing capability in digital circuitry may be utilized for configuring the limiter. This allows the configuration of the limiter without increasing the cost. The maximum amplitude of the limiter, however, is restricted if there is an analog circuit based signal level adjuster. Since the signal level adjuster adjusts signal levels, the maximum level of amplitude changes depends on the maximum amplitude of the signal level adjuster.
For example, if the restriction level of the limiter is set according to the 0 dB attenuation level of the signal level adjuster, the maximum level of the signal level adjuster may fall by 10 dB when the attenuation level of the signal level adjuster is set to xe2x88x9210 dB compared to when the attenuation level of the signal level adjuster is 0 dB. Therefore, the amplitude of the output of the signal level adjuster may be unnecessarily limited if the attenuation level in the limiter is set too high.
The present invention provides a sound processing circuit which may solve a range of problems related to the distribution of low frequency signal components as a result of the introduction of the aforementioned new multichannel recording and reproduction systems.
The sound processing circuit of the present invention filters the low frequency signal components in digital audio signals of m (mxe2x89xa6n) channels out of one exclusive channel for low frequency signals and n (n greater than 1) multiple independent channels, and outputs filtered low frequency signal components from a low frequency signal channel. The sound processing circuit of the present invention comprises m high-pass filters which receive m digital audio signals from the m specified channels, and allow to pass through signal components in higher bands than a cut-off frequency fc; m first coefficient multipliers which receive and multiply digital audio signals of the m specified channels by a multiplication coefficient ai (0 less than ai less than 1); a second coefficient multiplier which receives and multiplies the digital audio signals of the low frequency signals of the exclusive channel, by a multiplication coefficient aL (0 less than aL less than 1); an adder for adding each output of m first coefficient multipliers and the output of a second coefficient multiplier to produce synthetic audio signals; and a low pass filter which receives the synthetic audio signals from the adder, and allows to pass through signal components in lower bands than the cut-off frequency fc.
In the above configuration, low frequency signal components are filtered from input multichannel audio signals by a digital unit, and filtered low frequency signal components are distributed by an analog unit.
Using the above configuration, most of the complicated circuits required for distributing low frequency signal components in multichannel audio signals can be realized with digital circuits which facilitate configuration and control. In addition, a process for adding low frequency signal components to channels to which low frequency signal components are distributed can be realized with analog circuits which facilitate the assurance of an amplitude margin. This facilitates configuration and control of hardware while maintaining good sound quality by securing a sufficient amplitude margin.